<div dir="ltr"><div><div><div>Hi Jon , <br><br></div>So the fmtp that goes to the carrier is the same as the one sent by Janus ? <br></div>Try : <param name="inbound-codec-negotiation" value="greedy"/> on the SIP profile. I assume you have this set to "generous".<br></div><div>Also please use FS master or the latest release.<br></div><div>If it still does not work please open a FS jira. <br></div><div>You see the fmtp being changed with the XML values when you originate the call through fs_cli, right ? <br></div><div><br>Cheers,<br></div>Dragos<br><div><div><div><br></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jul 12, 2017 at 3:33 PM, Jonathan Hunter <span dir="ltr"><<a href="mailto:hunterj91@hotmail.com" target="_blank">hunterj91@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div id="m_1230162313274624816divtagdefaultwrapper" style="font-size:12pt;color:#000000;font-family:Calibri,Helvetica,sans-serif" dir="ltr">
<p><span style="font-size:12pt">Hi Dragos,</span></p>
<p><span style="font-size:12pt"><br>
</span></p>
<p><span style="font-size:12pt">Thank you for the fast reply!</span></p>
<p><span style="font-size:12pt"><br>
</span></p>
<p><span style="font-size:12pt">You are correct in terms of call flow.</span></p>
<p><span style="font-size:12pt"><br>
</span></p>
<p><span style="font-size:12pt">I am originating from a WebRTC client on Janus, which then sends a SIP invite into FreeSWITCH , I then reply to that to set it to 8K, narrowband, this works perfectly !
<span>😊</span></span><br>
</p>
<p><span style="font-size:12pt"><span><br>
</span></span></p>
<p><span style="font-size:12pt"><span>However I then bridge the call out to a carrier, and I want to offer in the SDP the maxplaybackrate=8000, however I cant seem to do this, and this is with and without late-negotiation enabled.</span></span></p>
<p><span style="font-size:12pt"><span><br>
</span></span></p>
<p><span style="font-size:12pt"><span>Am I missing something or do you have any debug suggestions for me please?</span></span></p>
<p><span style="font-size:12pt"><span><br>
</span></span></p>
<p><span style="font-size:12pt"><span>Many  thanks</span></span></p>
<p><span style="font-size:12pt"><span><br>
</span></span></p>
<p><span style="font-size:12pt"><span>Jon</span></span></p>
<div style="color:rgb(0,0,0)">
<div>
<div dir="ltr"><div><div class="h5"><br>
<div class="gmail_quote">---------- Forwarded message ----------<br>
From: <b class="gmail_sendername">Oancea, Dragos</b> <span dir="ltr"><<a href="mailto:dragos.oancea@vonage.com" target="_blank">dragos.oancea@vonage.com</a>></span><br>
Date: Wed, Jul 12, 2017 at 1:14 PM<br>
Subject: Re: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit<br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.<wbr>freeswitch.org</a>><br>
<br>
<br>
<div dir="ltr">Hi Jon,<br>
<div><br>
If both sides follow the RFC , then you should be able to get narrowband end to end.<br>
</div>
<div>How do you originate the call ? <br>
</div>
<div>Sounds like you originate a call from a device and then you bridge to the callee, but you have inbound-late-negotiation on, so the OPUS config XML is not loaded, hence fmtp cannot be changed according to config.<br>
<br>
</div>
<div>Regards,<br>
</div>
<div>Dragos<br>
</div>
<div>Â <br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">
<div>
<div class="m_1230162313274624816h5">On Wed, Jul 12, 2017 at 1:04 PM, Jonathan Hunter <span dir="ltr">
<<a href="mailto:hunterj91@hotmail.com" target="_blank">hunterj91@hotmail.com</a>></span> wrote:<br>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div class="m_1230162313274624816h5">
<div dir="ltr">
<div id="m_1230162313274624816m_-4133567141997644636m_1589231063622866357divtagdefaultwrapper" dir="ltr" style="font-size:12pt;color:#000000;font-family:Calibri,Helvetica,sans-serif">
<p></p>
<div>Hi Guys,</div>
<div><br>
</div>
<div>We are currently testing OPUS and Im looking to lock down things to a maxplaybackrate of 8000HZ.</div>
<div><br>
</div>
<div>This works fine if FreeSWITCH is the receiver (Offer Answer), however if I then send an invite out from FreeSWITCH (bridge the same call)it doesnt seem to add the  fmtp maxplaybackrate=8000 paramater.</div>
<div><br>
</div>
<div>As when FreeSWITCH answers I see;</div>
<div><br>
</div>
<div>Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; minptime=10</div>
<div><br>
</div>
<div>However when I send an Invite offer out I see;</div>
<div><br>
</div>
<div>Â Media Attribute (a): rtpmap:111 opus/48000/2</div>
<div>Â Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1</div>
<div><br>
</div>
<div><br>
</div>
<div>The OPUS module document seems to suggest we can add this, and if so how can I achieve this, as I have my opus.conf set to;</div>
<div><br>
</div>
<div><param name="use-vbr" value="1"/></div>
<div><param name="use-dtx" value="0"/></div>
<div><param name="complexity" value="10"/></div>
<div><param name="maxaveragebitrate" value="14400"/></div>
<div><param name="maxplaybackrate" value="8000"/></div>
<div><param name="packet-loss-percent" value="15"/></div>
<div><param name="keep-fec-enabled" value="1"/></div>
<div><param name="use-jb-lookahead" value="1"/></div>
<div><param name="advertise-useinbandfec" value="1"/></div>
<div><br>
</div>
<div><br>
</div>
<div>And I also set the SIP profile codec list and absolute codec offer to OPUS@8000h, and as I say works great when we answer an offer, but not when we originate.</div>
<div><br>
</div>
<div>Am I misconfiguring or is this not possible?</div>
<div><br>
</div>
<div>I know the RFC states the parameter "maxplaybackrate" is a unidirectional receive-only parameter that reflects limitations of the local receiver, however mod_opus documentation seems to suggest we can set it in the offer, so I should be able to add as;</div>
<div><br>
</div>
<div>maxplaybackrate=8000</div>
<div><br>
</div>
<div>Is this possible? I just want to ensure I get narrowband end to end.</div>
<div><br>
</div>
<div>Many thanks</div>
<div><br>
</div>
<div>Jon</div>
<br>
<p></p>
</div>
</div>
<br>
</div>
</div>
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-- <br>
<div class="m_1230162313274624816gmail_signature">
<div dir="ltr">
<div>Jonathan Hunter</div>
Technical Director /Telephony Developer
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